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<div class="title">Introduction </div>  </div>
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<div class="textblock"><h1><a class="anchor" id="SCOPE"></a>
Scope</h1>
<p>This document describes the high-level application interface and usage of the ISO/MPEG-2/4 AAC Decoder library developed by the Fraunhofer Institute for Integrated Circuits (IIS). Depending on the library configuration, decoding of AAC-LC (Low-Complexity), HE-AAC (High-Efficiency AAC v1 and v2), AAC-LD (Low-Delay) and AAC-ELD (Enhanced Low-Delay) is implemented.</p>
<p>All references to SBR (Spectral Band Replication) are only applicable to HE-AAC and AAC-ELD configurations of the FDK library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2 decoder configuration of the library.</p>
<h1><a class="anchor" id="DecoderBasics"></a>
Decoder Basics</h1>
<p>This document can only give a rough overview about the ISO/MPEG-2, ISO/MPEG-4 AAC audio and MPEG-D USAC coding standards. To understand all details referenced in this document, you are encouraged to read the following documents.</p>
<ul>
<li>ISO/IEC 13818-7 (MPEG-2 AAC) Standard, defines the syntax of MPEG-2 AAC audio bitstreams.</li>
<li>ISO/IEC 14496-3 (MPEG-4 AAC, subpart 1 and 4) Standard, defines the syntax of MPEG-4 AAC audio bitstreams.</li>
<li>ISO/IEC 23003-3 (MPEG-D USAC), defines MPEG-D USAC unified speech and audio codec.</li>
<li>Lutzky, Schuller, Gayer, Kr&auml;mer, Wabnik, "A guideline to audio codec
delay", 116th AES Convention, May 8, 2004</li>
</ul>
<p>In short, MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is partitioned into overlapping time portions and transformed into frequency domain. The spectral components are then quantized and coded using a highly efficient coding scheme.<br  />
 Encoded MPEG-2 and MPEG-4 AAC audio bitstreams are composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the length of individual frames is not restricted to a fixed number of bytes, but can take any length between 1 and 768 bytes.</p>
<p>In addition to the above mentioned frequency domain coding mode, MPEG-D USAC also employs a time domain Algebraic Code-Excited Linear Prediction (ACELP) speech coder core. This operating mode is selected by the encoder in order to achieve the optimum audio quality for different content type. Several enhancements allow achieving higher quality at lower bit rates compared to MPEG-4 HE-AAC.</p>
<h1><a class="anchor" id="SCOPE"></a>
Scope</h1>
<p>This document describes the high-level application interface and usage of the FDK PCM audio mixdown module library developed by the Fraunhofer Institute for Integrated Circuits (IIS). Depending on the library configuration, the module can manipulate the number of audio channels of a given PCM signal. It can create for example a two channel stereo audio signal from a given multi-channel configuration (e.g. 5.1 channels). </p>
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